diff --git a/modules/audio_processing/aec/aec_core.cc b/modules/audio_processing/aec/aec_core.cc index 49971ff655..f0deddc2aa 100644 --- a/modules/audio_processing/aec/aec_core.cc +++ b/modules/audio_processing/aec/aec_core.cc @@ -36,34 +36,6 @@ extern "C" { #include "system_wrappers/include/metrics.h" namespace webrtc { -namespace { -enum class DelaySource { - kSystemDelay, // The delay values come from the OS. - kDelayAgnostic, // The delay values come from the DA-AEC. -}; - -constexpr int kMinDelayLogValue = -200; -constexpr int kMaxDelayLogValue = 200; -constexpr int kNumDelayLogBuckets = 100; - -void MaybeLogDelayAdjustment(int moved_ms, DelaySource source) { - if (moved_ms == 0) - return; - switch (source) { - case DelaySource::kSystemDelay: - RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecDelayAdjustmentMsSystemValue", - moved_ms, kMinDelayLogValue, kMaxDelayLogValue, - kNumDelayLogBuckets); - return; - case DelaySource::kDelayAgnostic: - RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecDelayAdjustmentMsAgnosticValue", - moved_ms, kMinDelayLogValue, kMaxDelayLogValue, - kNumDelayLogBuckets); - return; - } -} -} // namespace - // Buffer size (samples) static const size_t kBufferSizeBlocks = 250; // 1 second of audio in 16 kHz. @@ -1864,15 +1836,11 @@ void WebRtcAec_ProcessFrames(AecCore* aec, // rounding, like -16. int move_elements = (aec->knownDelay - knownDelay - 32) / PART_LEN; int moved_elements = aec->farend_block_buffer_.AdjustSize(move_elements); - MaybeLogDelayAdjustment(moved_elements * (aec->sampFreq == 8000 ? 8 : 4), - DelaySource::kSystemDelay); aec->knownDelay -= moved_elements * PART_LEN; } else { // 2 b) Apply signal based delay correction. int move_elements = SignalBasedDelayCorrection(aec); int moved_elements = aec->farend_block_buffer_.AdjustSize(move_elements); - MaybeLogDelayAdjustment(moved_elements * (aec->sampFreq == 8000 ? 8 : 4), - DelaySource::kDelayAgnostic); int far_near_buffer_diff = aec->farend_block_buffer_.Size() - (aec->nearend_buffer_size + FRAME_LEN) / PART_LEN;